diff --git a/OpenMPT/build/android_ndk/Android-minimp3-stbvorbis.mk b/OpenMPT/build/android_ndk/Android-minimp3-stbvorbis.mk
index 7ec6552..05ad6fb 100644
--- a/OpenMPT/build/android_ndk/Android-minimp3-stbvorbis.mk
+++ b/OpenMPT/build/android_ndk/Android-minimp3-stbvorbis.mk
@@ -74,6 +74,7 @@ LOCAL_SRC_FILES := \
 	soundlib/Load_ptm.cpp \
 	soundlib/Load_s3m.cpp \
 	soundlib/Load_stm.cpp \
+	soundlib/Load_stp.cpp \
 	soundlib/Load_ult.cpp \
 	soundlib/Load_umx.cpp \
 	soundlib/Load_wav.cpp \
diff --git a/OpenMPT/build/android_ndk/Android-mpg123-vorbis.mk b/OpenMPT/build/android_ndk/Android-mpg123-vorbis.mk
index 0c09b5e..456aa21 100644
--- a/OpenMPT/build/android_ndk/Android-mpg123-vorbis.mk
+++ b/OpenMPT/build/android_ndk/Android-mpg123-vorbis.mk
@@ -72,6 +72,7 @@ LOCAL_SRC_FILES := \
 	soundlib/Load_ptm.cpp \
 	soundlib/Load_s3m.cpp \
 	soundlib/Load_stm.cpp \
+	soundlib/Load_stp.cpp \
 	soundlib/Load_ult.cpp \
 	soundlib/Load_umx.cpp \
 	soundlib/Load_wav.cpp \
diff --git a/OpenMPT/build/android_ndk/Android-unmo3.mk b/OpenMPT/build/android_ndk/Android-unmo3.mk
index f28d829..aa56d61 100644
--- a/OpenMPT/build/android_ndk/Android-unmo3.mk
+++ b/OpenMPT/build/android_ndk/Android-unmo3.mk
@@ -105,6 +105,7 @@ LOCAL_SRC_FILES := \
 	soundlib/Load_ptm.cpp \
 	soundlib/Load_s3m.cpp \
 	soundlib/Load_stm.cpp \
+	soundlib/Load_stp.cpp \
 	soundlib/Load_ult.cpp \
 	soundlib/Load_umx.cpp \
 	soundlib/Load_wav.cpp \
diff --git a/OpenMPT/build/android_ndk/Android.mk b/OpenMPT/build/android_ndk/Android.mk
index 6ef628b..6643057 100644
--- a/OpenMPT/build/android_ndk/Android.mk
+++ b/OpenMPT/build/android_ndk/Android.mk
@@ -73,6 +73,7 @@ LOCAL_SRC_FILES := \
 	soundlib/Load_ptm.cpp \
 	soundlib/Load_s3m.cpp \
 	soundlib/Load_stm.cpp \
+	soundlib/Load_stp.cpp \
 	soundlib/Load_ult.cpp \
 	soundlib/Load_umx.cpp \
 	soundlib/Load_wav.cpp \
diff --git a/OpenMPT/build/autotools/Makefile.am b/OpenMPT/build/autotools/Makefile.am
index 86dbdd2..ea60f94 100644
--- a/OpenMPT/build/autotools/Makefile.am
+++ b/OpenMPT/build/autotools/Makefile.am
@@ -185,6 +185,7 @@ libopenmpt_la_SOURCES += soundlib/Load_psm.cpp
 libopenmpt_la_SOURCES += soundlib/Load_ptm.cpp
 libopenmpt_la_SOURCES += soundlib/Load_s3m.cpp
 libopenmpt_la_SOURCES += soundlib/Load_stm.cpp
+libopenmpt_la_SOURCES += soundlib/Load_stp.cpp
 libopenmpt_la_SOURCES += soundlib/Load_ult.cpp
 libopenmpt_la_SOURCES += soundlib/Load_umx.cpp
 libopenmpt_la_SOURCES += soundlib/Load_wav.cpp
@@ -418,6 +419,7 @@ libopenmpttest_SOURCES += soundlib/Load_psm.cpp
 libopenmpttest_SOURCES += soundlib/Load_ptm.cpp
 libopenmpttest_SOURCES += soundlib/Load_s3m.cpp
 libopenmpttest_SOURCES += soundlib/Load_stm.cpp
+libopenmpttest_SOURCES += soundlib/Load_stp.cpp
 libopenmpttest_SOURCES += soundlib/Load_ult.cpp
 libopenmpttest_SOURCES += soundlib/Load_umx.cpp
 libopenmpttest_SOURCES += soundlib/Load_wav.cpp
diff --git a/OpenMPT/libopenmpt/foo_openmpt.cpp b/OpenMPT/libopenmpt/foo_openmpt.cpp
index ac7ce34..8947cdf 100644
--- a/OpenMPT/libopenmpt/foo_openmpt.cpp
+++ b/OpenMPT/libopenmpt/foo_openmpt.cpp
@@ -319,6 +319,7 @@ DECLARE_FILE_TYPE("OpenMPT compatible module files",
 	"*.imf" ";"
 	"*.j2b" ";"
 	"*.plm" ";"
+	"*.stp" ";"
 	"*.gdm" ";"
 	"*.umx" ";"
 	"*.mo3" ";"
diff --git a/OpenMPT/soundlib/Load_stp.cpp b/OpenMPT/soundlib/Load_stp.cpp
new file mode 100644
index 0000000..2a054ee
--- /dev/null
+++ b/OpenMPT/soundlib/Load_stp.cpp
@@ -0,0 +1,945 @@
+/*
+ * Load_stp.cpp
+ * ------------
+ * Purpose: STP (Soundtracker Pro II) module loader
+ * Notes  : a few exotic effects aren't supported.
+ *          multiple sample loops are supported, but only the first 10 can be used as cue points
+ *          (with 16xx and 18xx).
+ *          fractional speed values and combined auto effects are handled whenever possible,
+ *          but some effects may be omitted (and there may be tempo accuracy issues).
+ * Authors: Devin Acker
+ *          OpenMPT Devs
+ * The OpenMPT source code is released under the BSD license. Read LICENSE for more details.
+ *
+ * Wisdom from the Soundtracker Pro II manual:
+ * "To create shorter patterns, simply create shorter patterns."
+ */
+ 
+#include "stdafx.h"
+#include "Loaders.h"
+#include "Tables.h"
+
+OPENMPT_NAMESPACE_BEGIN
+
+#ifdef NEEDS_PRAGMA_PACK
+#pragma pack(push, 1)
+#endif
+
+// File header (except for "STP3" magic)
+struct PACKED STPFileHeader {
+	uint16 version;
+	uint8  numOrders;
+	uint8  patternLength;
+	uint8  orderList[128];
+	uint16 speed;
+	uint16 speedFrac;
+	uint16 timerCount;
+	uint16 flags;
+	uint32 reserved;
+	uint16 midiCount; // always 50
+	uint8  midi[50];
+	uint16 numSamples;
+	uint16 sampleStructSize;
+	
+	void ConvertEndianness()
+	{
+		SwapBytesBE(version);
+		SwapBytesBE(speed);
+		SwapBytesBE(speedFrac);
+		SwapBytesBE(timerCount);
+		SwapBytesBE(flags);
+		SwapBytesBE(midiCount);
+		SwapBytesBE(numSamples);
+		SwapBytesBE(sampleStructSize);
+	}
+};
+
+STATIC_ASSERT(sizeof(STPFileHeader) == 200);
+
+// Sample header (versions 0 and 1)
+struct PACKED STPSampleHeaderOld {
+	char   pathName[31];
+	uint8  flags;
+	char   fileName[30];
+	uint32 length;
+	uint8  volume;
+	uint8  reserved1;
+	uint32 loopStart;
+	uint32 loopLength;
+	uint16 defaultCmd;
+	uint32 reserved2;
+	
+	void ConvertEndianness()
+	{
+		SwapBytesBE(length);
+		SwapBytesBE(loopStart);
+		SwapBytesBE(loopLength);
+		SwapBytesBE(defaultCmd);
+	}
+	
+	void ConvertToMPT(ModSample &mptSmp) const
+	{
+		mptSmp.Initialize(MOD_TYPE_MOD);
+		mptSmp.nLength = length;
+		mptSmp.nVolume = 4 * MIN(volume, 64);
+		
+		mptSmp.nLoopStart = loopStart;
+		mptSmp.nLoopEnd = loopStart + loopLength;
+
+		if(mptSmp.nLoopStart >= mptSmp.nLength)
+		{
+			mptSmp.nLoopStart = mptSmp.nLength - 1;
+		}
+		if(mptSmp.nLoopEnd > mptSmp.nLength)
+		{
+			mptSmp.nLoopEnd = mptSmp.nLength;
+		}
+		
+		if(mptSmp.nLoopStart > mptSmp.nLoopEnd)
+		{
+			mptSmp.nLoopStart = 0;
+			mptSmp.nLoopEnd = 0;
+		}
+		else if(mptSmp.nLoopEnd > mptSmp.nLoopStart)
+		{
+			mptSmp.uFlags.set(CHN_LOOP);
+		}
+	}
+	
+	void Read(FileReader &file) {
+		file.ReadConvertEndianness(*this);
+		pathName[30] = '\0';
+	}
+};
+
+STATIC_ASSERT(sizeof(STPSampleHeaderOld) == 82);
+
+#ifdef NEEDS_PRAGMA_PACK
+#pragma pack(pop)
+#endif
+
+// Sample header (version 2), not packed due to variable string length and alignment in file
+struct STPSampleHeader {
+	char   pathName[256];
+	uint8  flags;
+	char   fileName[30];
+	uint32 length;
+	uint8  volume;
+	uint8  reserved1;
+	uint32 loopStart;
+	uint32 loopLength;
+	uint16 defaultCmd;
+	uint16 defaultPeriod;
+	uint8  finetune;
+	uint8  reserved2;
+	
+	void ConvertToMPT(ModSample &mptSmp) const
+	{
+		mptSmp.Initialize(MOD_TYPE_MOD);
+		mptSmp.nLength = length;
+		mptSmp.nFineTune = static_cast<int8>(finetune << 3);
+		mptSmp.nVolume = 4 * MIN(volume, 64);
+		
+		mptSmp.nLoopStart = loopStart;
+		mptSmp.nLoopEnd = loopStart + loopLength;
+
+		if(mptSmp.nLoopStart >= mptSmp.nLength)
+		{
+			mptSmp.nLoopStart = mptSmp.nLength - 1;
+		}
+		if(mptSmp.nLoopEnd > mptSmp.nLength)
+		{
+			mptSmp.nLoopEnd = mptSmp.nLength;
+		}
+		
+		if(mptSmp.nLoopStart > mptSmp.nLoopEnd)
+		{
+			mptSmp.nLoopStart = 0;
+			mptSmp.nLoopEnd = 0;
+		}
+		else if(mptSmp.nLoopEnd > mptSmp.nLoopStart)
+		{
+			mptSmp.cues[0] = mptSmp.nLoopStart;
+			mptSmp.uFlags.set(CHN_LOOP);
+		}
+	}
+	
+	void Read(FileReader &file) {
+		std::string str;
+		
+		file.ReadNullString(str, 255);
+		std::strncpy(pathName, str.c_str(), 255);
+		pathName[255] = '\0';
+		
+		flags = file.ReadUint8();
+		
+		file.ReadNullString(str, 29);
+		std::strncpy(fileName, str.c_str(), 29);
+		pathName[29] = '\0';
+		
+		// seek to even boundary
+		if(file.GetPosition() & 1)
+			file.Skip(1);
+		
+		length = file.ReadUint32BE();
+		volume = file.ReadUint8();
+		reserved1 = file.ReadUint8();
+		
+		loopStart = file.ReadUint32BE();
+		loopLength = file.ReadUint32BE();
+		
+		defaultCmd = file.ReadUint16BE();
+		defaultPeriod = file.ReadUint16BE();
+		
+		finetune = file.ReadUint8();
+		reserved2 = file.ReadUint8();
+	}
+};
+
+struct STPLoopInfo {
+	SmpLength loopStart;
+	SmpLength loopLength;
+	SAMPLEINDEX looped;
+	SAMPLEINDEX nonLooped;
+};
+
+typedef std::vector<STPLoopInfo> STPLoopList;
+
+template <typename T>
+static void ReadSample(FileReader &file, T &sampleHeader, ModSample &sample, char (&sampleName)[MAX_SAMPLENAME])
+//--------------------------------------------------------------------------------------------------------------
+{
+	sampleHeader.Read(file);
+	sampleHeader.ConvertToMPT(sample);
+
+	mpt::String::Read<mpt::String::spacePadded>(sampleName, sampleHeader.fileName);
+	// Get rid of weird characters in sample names.
+	for(size_t i = 0; i < CountOf(sampleName); i++)
+	{
+		if(sampleName[i] > 0 && sampleName[i] < ' ')
+		{
+			sampleName[i] = ' ';
+		}
+	}
+}
+
+static TEMPO ConvertTempo(uint16 ciaSpeed)
+//----------------------------------------
+{
+	// 3546 is the resulting CIA timer value when using 4F7D (tempo 125 bpm) command in STProII
+	return TEMPO(125.0 * (3546.0 / ciaSpeed));
+}
+
+static void ConvertLoopSlice(ModSample &src, ModSample &dest, SmpLength start, SmpLength len, bool loop)
+//------------------------------------------------------------------------------------------------------
+{
+	if(!src.HasSampleData()) return;
+	
+	dest.FreeSample();
+	
+	dest = src;
+	dest.nLength = len;
+	dest.pSample = nullptr;
+	
+	if(dest.AllocateSample() != len)
+	{
+		dest.FreeSample();
+		return;
+	}
+	
+	// only preserve cue points if the target sample length is the same
+	if(len != src.nLength)
+		memset(dest.cues, 0, sizeof(dest.cues));
+	
+	memcpy(dest.pSample8, src.pSample8 + start, len);
+	if(loop)
+	{
+		dest.nLoopStart = 0;
+		dest.nLoopEnd = len;
+		dest.uFlags.set(CHN_LOOP);
+	} else
+	{
+		dest.nLoopStart = 0;
+		dest.nLoopEnd = 0;
+		dest.uFlags.reset(CHN_LOOP);
+	}
+}
+
+static void ConvertLoopSequence(ModSample &smp, STPLoopList &loopList)
+//--------------------------------------------------------------------
+{
+	// This should only modify a sample if it has more than one loop
+	// (otherwise, it behaves like a normal sample loop)
+	if(!smp.HasSampleData() || loopList.size() < 2) return;
+	
+	ModSample newSmp = smp;
+	newSmp.nLength = 0;
+	newSmp.pSample = nullptr;
+	
+	SAMPLEINDEX numLoops = loopList.size();
+	
+	// get the total length of the sample after combining all looped sections
+	for(int i = 0; i < numLoops; i++)
+	{
+		STPLoopInfo &info = loopList[i];
+		
+		// if adding this loop would cause the sample length to exceed maximum, 
+		// then limit and bail out
+		if((newSmp.nLength + info.loopLength > MAX_SAMPLE_LENGTH) || 
+		   (info.loopLength > MAX_SAMPLE_LENGTH) ||
+		   (info.loopStart + info.loopLength > smp.nLength))
+		{
+			numLoops = i;
+			break;
+		}
+		
+		newSmp.nLength += info.loopLength;
+	}
+	
+	if(newSmp.AllocateSample() != newSmp.nLength)
+	{
+		newSmp.FreeSample();
+		return;
+	}
+	
+	// start copying the looped sample data parts
+	SmpLength start = 0;
+		
+	for(int i = 0; i < numLoops; i++)
+	{
+		STPLoopInfo &info = loopList[i];
+		
+		memcpy(newSmp.pSample8 + start, smp.pSample8 + info.loopStart, info.loopLength);
+		
+		// update loop info based on position in edited sample
+		info.loopStart = start;
+		if(i > 0 && i < 10)
+		{
+			newSmp.cues[i-1] = start;
+		}
+		start += info.loopLength;
+	}
+	
+	// replace old sample with new one
+	smp.FreeSample();
+	smp = newSmp;
+	
+	smp.nLoopStart = 0;
+	smp.nLoopEnd = smp.nLength;
+	smp.uFlags.set(CHN_LOOP);
+}
+
+bool CSoundFile::ReadSTP(FileReader &file, ModLoadingFlags loadFlags)
+//-------------------------------------------------------------------
+{
+	file.Rewind();
+	if(!file.ReadMagic("STP3"))
+		return false;
+
+	STPFileHeader fileHeader;
+	file.ReadConvertEndianness(fileHeader);
+	if(fileHeader.version > 2 || 
+	   fileHeader.numOrders > 128 || fileHeader.numSamples > MAX_SAMPLES ||
+	   fileHeader.midiCount != 50)
+		return false;
+	
+	if(loadFlags == onlyVerifyHeader)
+		return true;
+		
+	InitializeGlobals(MOD_TYPE_STP);
+	
+	m_nChannels = 4;
+	m_nSamples = 0;
+	m_nInstruments = 0;
+	
+	m_nDefaultSpeed = fileHeader.speed;
+	m_nDefaultTempo = ConvertTempo(fileHeader.timerCount);
+	
+	m_nMinPeriod = 14 * 4;
+	m_nMaxPeriod = 3424 * 4;
+	
+	Order.ReadFromArray(fileHeader.orderList, fileHeader.numOrders);
+
+	std::vector<STPLoopList> loopInfo;
+	// non-looped versions of samples with loopes (when needed)
+	std::vector<SAMPLEINDEX> nonLooped;
+
+	// Load sample headers
+	SAMPLEINDEX samplesInFile = 0;
+	
+	for(SAMPLEINDEX smp = 1; smp <= fileHeader.numSamples; smp++)
+	{
+		// this is 1-based the same as smp
+		SAMPLEINDEX actualSmp = file.ReadUint16BE();
+		if(actualSmp > MAX_SAMPLES)
+			return false;
+		
+		samplesInFile = m_nSamples = MAX(m_nSamples, actualSmp);
+		
+		ModSample &thisSmp = Samples[actualSmp];
+		
+		if(fileHeader.version == 2)
+		{
+			STPSampleHeader sampleHeader;
+			uint32 headerSize = file.ReadUint32BE();
+			
+			ReadSample(file, sampleHeader, thisSmp, m_szNames[actualSmp]);
+			// TODO: verify string lengths against headerSize?
+			(void)headerSize;
+		} else
+		{
+			STPSampleHeaderOld sampleHeaderOld;
+			ReadSample(file, sampleHeaderOld, thisSmp, m_szNames[actualSmp]);
+		}
+		
+		STPLoopList loopList;
+		
+		if(fileHeader.version >= 1)
+		{
+			uint16 numLoops = file.ReadUint16BE();
+			
+			STPLoopInfo loop;
+			loop.looped = loop.nonLooped = 0;
+				
+			if(numLoops == 0 && thisSmp.uFlags[CHN_LOOP])
+			{
+				loop.loopStart  = thisSmp.nLoopStart;
+				loop.loopLength = thisSmp.nLoopEnd - thisSmp.nLoopStart;
+				loopList.push_back(loop);
+				
+			} else for(SAMPLEINDEX i = 0; i < numLoops; i++)
+			{
+				loop.loopStart  = file.ReadUint32BE();
+				loop.loopLength = file.ReadUint32BE();
+				loopList.push_back(loop);
+			}
+		}
+		
+		nonLooped.resize(actualSmp);
+		loopInfo.resize(actualSmp);
+		loopInfo[actualSmp - 1] = loopList;
+	}
+
+	// Load patterns
+	uint16 numPatterns = 128;
+	if (fileHeader.version == 0)
+		numPatterns = file.ReadUint16BE();
+	
+	uint16 patternLength = fileHeader.patternLength;
+	uint16 channels = 4;
+	
+	uint8 globalVolSlide = 0;
+	std::vector<uint8> autoFinePorta;
+	std::vector<uint8> autoPortaUp;
+	std::vector<uint8> autoPortaDown;
+	std::vector<uint8> autoVolSlide;
+	std::vector<uint8> autoVibrato;
+	std::vector<uint8> vibratoMem;
+	std::vector<uint8> autoTremolo;
+	std::vector<uint8> autoTonePorta;
+	std::vector<uint8> tonePortaMem;
+	
+	for(PATTERNINDEX pat = 0; pat < numPatterns; pat++)
+	{
+		PATTERNINDEX actualPat = pat;
+		
+		if(fileHeader.version > 0)
+		{
+			actualPat = file.ReadUint16BE();
+			if(actualPat == 0xFFFF)
+				break;
+			
+			patternLength = file.ReadUint16BE();
+			channels = file.ReadUint16BE();
+			
+			m_nChannels = MIN(m_nChannels, channels);
+		}
+		
+		if(!(loadFlags & loadPatternData) || !Patterns.Insert(actualPat, patternLength))
+		{
+			file.Skip(channels * patternLength * 4);
+			continue;
+		}
+		
+		autoFinePorta.resize(channels, 0);
+		autoPortaUp.resize(channels, 0);
+		autoPortaDown.resize(channels, 0);
+		autoVolSlide.resize(channels, 0);
+		autoVibrato.resize(channels, 0);
+		vibratoMem.resize(channels, 0);
+		autoTremolo.resize(channels, 0);
+		autoTonePorta.resize(channels, 0);
+		tonePortaMem.resize(channels, 0);
+		
+		for(ROWINDEX row = 0; row < patternLength; row++)
+		{
+			PatternRow rowBase = Patterns[actualPat].GetpModCommand(row, 0);
+
+			bool didGlobalVolSlide = false;
+			bool shouldDelay = false;
+			
+			// if a fractional speed value is in use then determine if we should stick a fine pattern delay somewhere
+			switch(fileHeader.speedFrac & 3)
+			{
+			// 1/4
+			case 1: shouldDelay = (row & 3) == 0; break;
+			// 1/2
+			case 2: shouldDelay = (row & 1) == 0; break;
+			// 3/4
+			case 3: shouldDelay = (row & 3) != 3; break;
+			}
+			
+			for(CHANNELINDEX chn = 0; chn < channels; chn++)
+			{
+				ModCommand &m = rowBase[chn];
+				
+				m.instr   = file.ReadUint8();
+				m.note    = file.ReadUint8();
+				m.command = file.ReadUint8();
+				m.param   = file.ReadUint8();
+				
+				if(m.note)
+				{
+					m.note += 25;
+					
+					autoFinePorta[chn] = 0;
+					autoPortaUp[chn] = 0;
+					autoPortaDown[chn] = 0;
+					autoVolSlide[chn] = 0;
+					autoVibrato[chn] = vibratoMem[chn] = 0;
+					autoTremolo[chn] = 0;
+					autoTonePorta[chn] = tonePortaMem[chn] = 0;
+				}
+				
+				// this is a nibble-swapped param value used for auto fine volside
+				// and auto global fine volside
+				uint8 swap = (m.param >> 4) | (m.param << 4);
+				
+				if((m.command & 0xF0) == 0xF0)
+				{
+					m.param = Util::Round<uint8>(ConvertTempo(m.param | (((uint16)m.command & 0xF) << 8)).ToDouble());
+					m.command = CMD_TEMPO;
+				} else switch(m.command)
+				{
+				case 0x00: // arpeggio
+					if(m.param)
+						m.command = CMD_ARPEGGIO; 
+					break;
+					
+				case 0x01: // portamento up
+					m.command = CMD_PORTAMENTOUP; 
+					break;
+					
+				case 0x02: // portamento down
+					m.command = CMD_PORTAMENTODOWN; 
+					break;
+				
+				case 0x03: // auto fine portamento up
+					autoFinePorta[chn] = 0x10 | MIN(m.param, 15);
+					autoPortaUp[chn] = 0;
+					autoPortaDown[chn] = 0;
+					autoTonePorta[chn] = 0;
+				
+					m.command = m.param = 0;
+					break;
+					
+				case 0x04: // auto fine portamento down
+					autoFinePorta[chn] = 0x20 | MIN(m.param, 15);
+					autoPortaUp[chn] = 0;
+					autoPortaDown[chn] = 0;
+					autoTonePorta[chn] = 0;
+				
+					m.command = m.param = 0;
+					break;
+				
+				case 0x05: // auto portamento up
+					autoFinePorta[chn] = 0;
+					autoPortaUp[chn] = m.param;
+					autoPortaDown[chn] = 0;
+					autoTonePorta[chn] = 0;
+				
+					m.command = m.param = 0;
+					break;
+				
+				case 0x06: // auto portamento down
+					autoFinePorta[chn] = 0;
+					autoPortaUp[chn] = 0;
+					autoPortaDown[chn] = m.param;
+					autoTonePorta[chn] = 0;
+				
+					m.command = m.param = 0;
+					break;
+				
+				case 0x07: // set global volume
+					m.command = CMD_GLOBALVOLUME;
+					globalVolSlide = 0;
+					break;
+				
+				case 0x08: // auto global fine volume slide
+					globalVolSlide = swap;
+					m.command = m.param = 0;
+					break;
+				
+				case 0x09: // fine portamento up
+					m.command = CMD_MODCMDEX;
+					m.param = 0x10 | MIN(m.param, 15);
+					break;
+					
+				case 0x0A: // fine portamento down
+					m.command = CMD_MODCMDEX;
+					m.param = 0x20 | MIN(m.param, 15);
+					break;
+				
+				case 0x0B: // auto fine volume slide
+					autoVolSlide[chn] = swap;
+					m.command = m.param = 0;
+					break;
+					
+				case 0x0C: // set volume
+					m.volcmd = VOLCMD_VOLUME;
+					m.vol = m.param;
+					autoVolSlide[chn] = 0;
+					m.command = m.param = 0;
+					break;
+					
+				case 0x0D: // volume slide (param is swapped compared to .mod)
+					if(m.param & 0xF0)
+					{
+						m.volcmd = VOLCMD_VOLSLIDEDOWN;
+						m.vol = m.param >> 4;
+					} else if(m.param & 0x0F)
+					{
+						m.volcmd = VOLCMD_VOLSLIDEUP;
+						m.vol = m.param & 0xF;
+					}
+					autoVolSlide[chn] = 0;
+					m.command = m.param = 0;
+					break;
+				
+				case 0x0E: // set filter (also uses opposite value compared to .mod)
+					m.command = CMD_MODCMDEX;
+					m.param = 1 ^ MIN(m.param, 1);
+					break;
+					
+				case 0x0F: // set speed
+					m.command = CMD_SPEED;
+					fileHeader.speedFrac = m.param & 0xF;
+					m.param >>= 4; 
+					break;
+				
+				case 0x10: // auto vibrato
+					autoVibrato[chn] = m.param;
+					vibratoMem[chn] = 0;
+					m.command = m.param = 0;
+					break;
+					
+				case 0x11: // auto tremolo
+					if(m.param & 0xF)
+						autoTremolo[chn] = m.param;
+					else
+						autoTremolo[chn] = 0;
+					m.command = m.param = 0;
+					break;
+				
+				case 0x12: // pattern break
+					m.command = CMD_PATTERNBREAK; 
+					break;
+				
+				case 0x13: // auto tone portamento
+					autoFinePorta[chn] = 0;
+					autoPortaUp[chn] = 0;
+					autoPortaDown[chn] = 0;
+					autoTonePorta[chn] = m.param;
+				
+					tonePortaMem[chn] = 0;
+					m.command = m.param = 0;
+					break;
+				
+				case 0x14: // position jump
+					m.command = CMD_POSITIONJUMP; 
+					break;
+				
+				case 0x16: // start loop sequence
+					if(m.instr && m.instr <= loopInfo.size())
+					{
+						STPLoopList& loopList = loopInfo[m.instr-1];
+					
+						m.param--;
+						if(m.param < MIN(9u, loopList.size()))
+						{
+							m.volcmd = VOLCMD_OFFSET;
+							m.vol = m.param;
+						}
+					}
+					
+					m.command = m.param = 0;
+					break;
+				
+				case 0x17: // play only loop nn
+					if(m.instr && m.instr <= loopInfo.size())
+					{
+						STPLoopList& loopList = loopInfo[m.instr-1];
+					
+						m.param--;
+						if (m.param < loopList.size())
+						{
+							if (!loopList[m.param].looped && m_nSamples < MAX_SAMPLES-1)
+								loopList[m.param].looped = ++m_nSamples;
+							m.instr = loopList[m.param].looped;
+						}
+					}
+					
+					m.command = m.param = 0;
+					break;
+				
+				case 0x18: // play sequence without loop
+					if(m.instr && m.instr <= loopInfo.size())
+					{
+						STPLoopList& loopList = loopInfo[m.instr-1];
+					
+						m.param--;
+						if(m.param < MIN(9u, loopList.size()))
+						{
+							m.volcmd = VOLCMD_OFFSET;
+							m.vol = m.param;
+						}
+						// switch to non-looped version of sample and create it if needed
+						if (!nonLooped[m.instr-1] && m_nSamples < MAX_SAMPLES-1)
+							nonLooped[m.instr-1] = ++m_nSamples;
+						m.instr = nonLooped[m.instr-1];
+					}
+					
+					m.command = m.param = 0;
+					break;
+					
+				case 0x19: // play only loop nn without loop
+					if(m.instr && m.instr <= loopInfo.size())
+					{
+						STPLoopList& loopList = loopInfo[m.instr-1];
+					
+						m.param--;
+						if (m.param < loopList.size())
+						{
+							if (!loopList[m.param].nonLooped && m_nSamples < MAX_SAMPLES-1)
+								loopList[m.param].nonLooped = ++m_nSamples;
+							m.instr = loopList[m.param].nonLooped;
+						}
+					}
+					
+					m.command = m.param = 0;
+					break;
+				
+				case 0x1D: // fine volume slide (nibble order also swapped)
+					m.command = CMD_VOLUMESLIDE;
+					m.param = swap;
+					if(m.param & 0xF0) // slide down
+						m.param |= 0x0F;
+					else if(m.param & 0x0F)
+						m.param |= 0xF0;
+					break;
+				
+				case 0x20: // "delayed fade"
+					// just behave like either a normal fade or a notecut
+					// depending on the speed
+					if(m.param & 0xF0)
+					{
+						autoVolSlide[chn] = m.param >> 4;
+						m.command = m.param = 0;
+					} else 
+					{
+						m.command = CMD_MODCMDEX;
+						m.param = 0xC0 | (m.param & 0xF);
+					}
+					break;
+				
+				case 0x21: // note delay
+					m.command = CMD_MODCMDEX;
+					m.param = 0xD0 | MIN(m.param, 15);
+					break;
+				
+				case 0x22: // retrigger note
+					m.command = CMD_RETRIG;
+					m.param = MIN(m.param, 15);
+					break;
+				
+				case 0x49: // set sample offset
+					m.command = CMD_OFFSET; 
+					break;
+				
+				case 0x4E: // other protracker commands (pattern loop / delay)
+					if((m.param & 0xF0) == 0x60 || (m.param & 0xF0) == 0xE0)
+						m.command = CMD_MODCMDEX;
+					else
+						m.command = m.param = 0;
+					break;
+				
+				case 0x4F: // set speed/tempo
+					if(m.param < 0x20)
+					{
+						m.command = CMD_SPEED;
+						fileHeader.speedFrac = 0;
+					} else
+					{
+						m.command = CMD_TEMPO;
+					}
+					break;
+				
+				default:
+					m.command = m.param = 0; 
+					break;
+				}
+				
+				bool didVolSlide = false;
+				
+				// try to put volume slide in volume command
+				if(autoVolSlide[chn] && !m.volcmd)
+				{
+					if(autoVolSlide[chn] & 0xF0)
+					{
+						m.volcmd = VOLCMD_FINEVOLUP;
+						m.vol = autoVolSlide[chn] >> 4;
+					} else
+					{
+						m.volcmd = VOLCMD_FINEVOLDOWN;
+						m.vol = autoVolSlide[chn] & 0xF;
+					}
+					didVolSlide = true;
+				}
+				
+				// try to place/combine all remaining running effects.
+				if(!m.command)
+				{
+					if(autoPortaUp[chn])
+					{
+						m.command = CMD_PORTAMENTOUP;
+						m.param = autoPortaUp[chn];
+						
+					} else if(autoPortaDown[chn])
+					{
+						m.command = CMD_PORTAMENTODOWN;
+						m.param = autoPortaDown[chn];
+						
+					} else if(autoFinePorta[chn])
+					{
+						m.command = CMD_MODCMDEX;
+						m.param = autoFinePorta[chn];
+						
+					} else if(autoTonePorta[chn])
+					{
+						m.command = CMD_TONEPORTAMENTO;
+						m.param = tonePortaMem[chn] = autoTonePorta[chn];
+						
+					} else if(autoVibrato[chn])
+					{
+						m.command = CMD_VIBRATO;
+						m.param = vibratoMem[chn] = autoVibrato[chn];
+						
+					} else if(!didVolSlide && autoVolSlide[chn])
+					{
+						m.command = CMD_VOLUMESLIDE;
+						m.param = autoVolSlide[chn];
+						// convert to a "fine" value by setting the other nibble to 0xF
+						if(m.param & 0x0F)
+							m.param |= 0xF0;
+						else if(m.param & 0xF0)
+							m.param |= 0x0F;
+						didVolSlide = true;
+						
+					} else if(autoTremolo[chn])
+					{
+						m.command = CMD_TREMOLO;
+						m.param = autoTremolo[chn];
+						
+					} else if(shouldDelay)
+					{
+						// insert a fine pattern delay here
+						m.command = CMD_S3MCMDEX;
+						m.param = 0x61;
+						shouldDelay = false;
+					
+					} else if(!didGlobalVolSlide && globalVolSlide)
+					{
+						m.command = CMD_GLOBALVOLSLIDE;
+						m.param = globalVolSlide;
+						// convert to a "fine" value by setting the other nibble to 0xF
+						if(m.param & 0x0F)
+							m.param |= 0xF0;
+						else if(m.param & 0xF0)
+							m.param |= 0x0F;
+				
+						didGlobalVolSlide = true;
+					}
+				}
+			}
+			
+			// TODO: create/use extra channels for global volslide/delay if needed
+		}
+	}
+
+	// after we know how many channels there really are...
+	m_nSamplePreAmp = 256 / m_nChannels;
+	// Setup channel pan positions and volume
+	SetupMODPanning(true);
+
+	// Skip over scripts and drumpad info
+	if(fileHeader.version > 0)
+	{
+		uint16 scriptNum;
+		uint32 length;
+		
+		while(file.CanRead(2))
+		{
+			scriptNum = file.ReadUint16BE();
+			if(scriptNum == 0xFFFF)
+				break;
+			
+			file.Skip(2);
+			length = file.ReadUint32BE();
+			file.Skip(length);
+		}
+		
+		// skip drumpad stuff
+		file.Skip(17*2);
+	}
+
+	// Reading samples
+	if(loadFlags & loadSampleData)
+	{
+		for(SAMPLEINDEX smp = 1; smp <= samplesInFile; smp++) if(Samples[smp].nLength)
+		{
+			SampleIO(
+				SampleIO::_8bit,
+				SampleIO::mono,
+				SampleIO::littleEndian,
+				SampleIO::signedPCM)
+				.ReadSample(Samples[smp], file);
+				
+			ConvertLoopSequence(Samples[smp], loopInfo[smp-1]);
+			
+			// make a non-looping duplicate of this sample if needed
+			if(nonLooped[smp-1])
+			{
+				ConvertLoopSlice(Samples[smp], Samples[nonLooped[smp-1]], 0, Samples[smp].nLength, false);
+			}
+			
+			for(SAMPLEINDEX loop = 0; loop < loopInfo[smp-1].size(); loop++)
+			{
+				STPLoopInfo &info = loopInfo[smp-1][loop];
+			
+				// make duplicate samples for this individual section if needed
+				if(info.looped)
+				{
+					ConvertLoopSlice(Samples[smp], Samples[info.looped], info.loopStart, info.loopLength, true);
+				}
+				if(info.nonLooped)
+				{
+					ConvertLoopSlice(Samples[smp], Samples[info.nonLooped], info.loopStart, info.loopLength, false);
+				}
+			}
+		}
+	}
+
+	return true;
+}
+
+OPENMPT_NAMESPACE_END
diff --git a/OpenMPT/soundlib/Snd_defs.h b/OpenMPT/soundlib/Snd_defs.h
index 687b0e8..ce5fa00 100644
--- a/OpenMPT/soundlib/Snd_defs.h
+++ b/OpenMPT/soundlib/Snd_defs.h
@@ -97,6 +97,7 @@ enum MODTYPE
 	MOD_TYPE_DIGI	= 0x8000000,
 	MOD_TYPE_UAX	= 0x10000000, // sampleset as module
 	MOD_TYPE_PLM	= 0x20000000,
+	MOD_TYPE_STP	= 0x40000000,
 };
 DECLARE_FLAGSET(MODTYPE)
 
diff --git a/OpenMPT/soundlib/Snd_fx.cpp b/OpenMPT/soundlib/Snd_fx.cpp
index 3542afb..7a482bc 100644
--- a/OpenMPT/soundlib/Snd_fx.cpp
+++ b/OpenMPT/soundlib/Snd_fx.cpp
@@ -3363,7 +3363,7 @@ void CSoundFile::PortamentoUp(CHANNELINDEX nChn, ModCommand::PARAM param, const
 	else
 		param = pChn->nOldPortaUpDown;
 
-	const bool doFineSlides = !doFinePortamentoAsRegular && !(GetType() & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_MT2 | MOD_TYPE_MED | MOD_TYPE_AMF0 | MOD_TYPE_DIGI));
+	const bool doFineSlides = !doFinePortamentoAsRegular && !(GetType() & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_MT2 | MOD_TYPE_MED | MOD_TYPE_AMF0 | MOD_TYPE_DIGI | MOD_TYPE_STP));
 
 	// Process MIDI pitch bend for instrument plugins
 	MidiPortamento(nChn, param, doFineSlides);
@@ -3422,7 +3422,7 @@ void CSoundFile::PortamentoDown(CHANNELINDEX nChn, ModCommand::PARAM param, cons
 	else
 		param = pChn->nOldPortaUpDown;
 
-	const bool doFineSlides = !doFinePortamentoAsRegular && !(GetType() & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_MT2 | MOD_TYPE_MED | MOD_TYPE_AMF0 | MOD_TYPE_DIGI));
+	const bool doFineSlides = !doFinePortamentoAsRegular && !(GetType() & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_MT2 | MOD_TYPE_MED | MOD_TYPE_AMF0 | MOD_TYPE_DIGI | MOD_TYPE_STP));
 
 	// Process MIDI pitch bend for instrument plugins
 	MidiPortamento(nChn, -static_cast<int>(param), doFineSlides);
@@ -3894,7 +3894,7 @@ void CSoundFile::VolumeSlide(ModChannel *pChn, ModCommand::PARAM param)
 	else
 		param = pChn->nOldVolumeSlide;
 
-	if((GetType() & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_MT2 | MOD_TYPE_MED | MOD_TYPE_DIGI)))
+	if((GetType() & (MOD_TYPE_MOD | MOD_TYPE_XM | MOD_TYPE_MT2 | MOD_TYPE_MED | MOD_TYPE_DIGI | MOD_TYPE_STP)))
 	{
 		// MOD / XM nibble priority
 		if((param & 0xF0) != 0)
@@ -5458,7 +5458,7 @@ uint32 CSoundFile::GetFreqFromPeriod(uint32 period, uint32 c5speed, int32 nPerio
 //------------------------------------------------------------------------------------------
 {
 	if (!period) return 0;
-	if (GetType() & (MOD_TYPE_MED|MOD_TYPE_MOD|MOD_TYPE_DIGI|MOD_TYPE_MTM|MOD_TYPE_AMF0))
+	if (GetType() & (MOD_TYPE_MED|MOD_TYPE_MOD|MOD_TYPE_DIGI|MOD_TYPE_MTM|MOD_TYPE_AMF0|MOD_TYPE_STP))
 	{
 		return ((3546895L * 4) << FREQ_FRACBITS) / period;
 	} else if (GetType() == MOD_TYPE_XM)
diff --git a/OpenMPT/soundlib/Sndfile.cpp b/OpenMPT/soundlib/Sndfile.cpp
index c463551..30df0c8 100644
--- a/OpenMPT/soundlib/Sndfile.cpp
+++ b/OpenMPT/soundlib/Sndfile.cpp
@@ -310,6 +310,7 @@ bool CSoundFile::Create(FileReader file, ModLoadingFlags loadFlags)
 		 && !ReadJ2B(file, loadFlags)
 		 && !ReadMO3(file, loadFlags)
 		 && !ReadPT36(file, loadFlags)
+		 && !ReadSTP(file, loadFlags)
 		 && !ReadMod(file, loadFlags)
 		 && !Read669(file, loadFlags)
 		 && !ReadICE(file, loadFlags)
@@ -1043,6 +1044,8 @@ MODTYPE CSoundFile::GetBestSaveFormat() const
 	case MOD_TYPE_AMF0:
 	case MOD_TYPE_DIGI:
 		return MOD_TYPE_MOD;
+	case MOD_TYPE_STP:
+		return MOD_TYPE_XM;
 	case MOD_TYPE_MED:
 		if(m_nDefaultTempo == TEMPO(125, 0) && m_nDefaultSpeed == 6 && !m_nInstruments)
 		{
diff --git a/OpenMPT/soundlib/Sndfile.h b/OpenMPT/soundlib/Sndfile.h
index 5b1f4d1..c3f8dee 100644
--- a/OpenMPT/soundlib/Sndfile.h
+++ b/OpenMPT/soundlib/Sndfile.h
@@ -670,6 +670,7 @@ public:
 	bool ReadDIGI(FileReader &file, ModLoadingFlags loadFlags = loadCompleteModule);
 	bool ReadPLM(FileReader &file, ModLoadingFlags loadFlags = loadCompleteModule);
 	bool ReadMID(FileReader &file, ModLoadingFlags loadFlags = loadCompleteModule);
+	bool ReadSTP(FileReader &file, ModLoadingFlags loadFlags = loadCompleteModule);
 
 	static std::vector<const char *> GetSupportedExtensions(bool otherFormats);
 	static mpt::Charset GetCharsetFromModType(MODTYPE modtype);
diff --git a/OpenMPT/soundlib/Tables.cpp b/OpenMPT/soundlib/Tables.cpp
index abf91a9..06250bd 100644
--- a/OpenMPT/soundlib/Tables.cpp
+++ b/OpenMPT/soundlib/Tables.cpp
@@ -87,6 +87,7 @@ static const ModFormatInfo modFormatInfo[] =
 	{ MOD_TYPE_IMF,		"Imago Orpheus",			"imf" },
 	{ MOD_TYPE_J2B,		"Galaxy Sound System",		"j2b" },
 	{ MOD_TYPE_PLM,		"Disorder Tracker 2",		"plm" },
+	{ MOD_TYPE_STP,		"Soundtracker Pro II",		"stp" },
 
 #ifndef NO_ARCHIVE_SUPPORT
 	// Compressed modules
@@ -147,6 +148,7 @@ static const ModCharsetInfo ModCharsetInfos[] =
 	{ MOD_TYPE_OKT , mpt::CharsetISO8859_1  },
 	{ MOD_TYPE_DBM , mpt::CharsetISO8859_1  },
 	{ MOD_TYPE_DIGI, mpt::CharsetISO8859_1  },
+	{ MOD_TYPE_STP,  mpt::CharsetISO8859_1  },
 	// Amiga // DOS
 	{ MOD_TYPE_MOD , mpt::CharsetISO8859_1  },
 	{ MOD_TYPE_MED , mpt::CharsetISO8859_1  },
